Telecommunications networks and other networks are increasing in both size and complexity in order to serve the growing demand for high speed communication links for the transfer of voice and/or data information. As these telecommunication networks approach capacity, alternate solutions or networks are sought to meet the demand for increasing network bandwidth.
Traditionally, voice calls are transported entirely over an end-to-end, circuit-switched (CS) Public Switched Telephone Network (PSTN). However, considerable attention has been directed toward the implementation of real-time communication across computer data (otherwise known as Internet Protocol, IP) networks, and particularly the ability to route voice traffic over these networks. Interest has also been raised in using Voice over IP (VoIP) solutions to facilitate voice communication between originating and terminating PSTN end points or from an originating PSTN point to an IP destination as such solutions substantially bypass the PSTN.
To facilitate call routing in such environments, originating and terminating switches can be connected to PSTN/IP gateways that belong to both the IP network and the PSTN. Based on the called number or other signaling indicator, the switches route certain calls through the IP gateways instead of the PSTN. In one typical example, Session Initiation Protocol (SIP) telephony is commonly used to establish the call set-up process. SIP is the predominant IETF standard and it is the chosen signaling protocol for upcoming 3GPP and 3GPP2 all-IP mobile networks.
The current approach for a CS to SIP call set-up includes a number of steps. In an IP network, call set-up (SIP INVITE) requests are addressed to SIP URLs (alphanumerical strings similar to email addresses that are associated with users rather than devices). Since SIP URLs cannot be dialed from a 12-key-pad of a typical telephone, a SIP user needs to get a phone number (pn) assigned in order to make him or her reachable from a CS network. In the CS network, pn is associated with one or a pool of telephony gateways. Accordingly, calls to pn are always routed to one of these associated gateways regardless of traffic levels at the associated gateway and actual location of the SIP end point. Next, the gateway either uses a pre-configured table that maps the phone numbers of its number range to SIP URLs, or it consults a server that stores these mappings in order to forward an incoming CS call request. Finally, the gateway converts the incoming CS call request to a SIP INVITE request with the SIP URL obtained earlier as the target address. This INVITE request is sent to the SIP network, where a SIP location service is used to map the personal SIP URL to a physical IP end point. After this address mapping in the SIP network, the call request is forwarded to the callee. The problem with this approach is that the telephony gateway is assigned without knowing the callee's current location. Specifically, a personal SIP URL does not contain routing information. As such, it has to be mapped to a physical IP address to complete the call set up. This mapping step is only done after the call has been routed to the SIP network (i.e., the call has been routed to the gateway associated with pn and then the physical IP endpoint is determined). As a consequence, optimal routing cannot be supported.